AMR multimode codec for coding speech signals having different degrees for robustness

ABSTRACT

A delay unit is provided to delay adapting modes of a codec if channel communication conditions are improving. A less robust mode having a higher intrinsic quality is selected only after some delay. The quality of the communication is measured to give quality values for successive measurement occasions. The delay unit comprises a plurality of memory cells in a memory for storing the quality values of the communication conditions of the channel for the most recent measurement occasions. A selector operates according to an algorithm for selecting one of the memory cells. The content of the selected cell is used to set the mode of the codec.

RELATED APPLICATIONS

This application claims priority and benefit from Swedish patentapplication No. 0102849-7, filed Aug. 22, 2001, the entire teachings ofwhich are incorporated herein by reference.

TECHNICAL FIELD

The present invention relates to adaptation of the modes of a codec usedin a digital telecommunication such as a mobile telecommunication systemin order to improve the performance of link adaptation and thus thequality of the transmitted speech. More specifically, the invention isrelated to adaptation of the modes of speech codecs according to the AMRstandard.

DESCRIPTION OF RELATED ART

Description of Systems Using AMR

In a cellular radio telecommunication system for digital transmissioninformation is transmitted between central stations and user stations.The total bit rate used in the transmissions from a user station to acentral station is in present advanced telephony systems fortransmission of speech information determined by the methods used forspeech coding, channel coding and modulation, and for a system based onTDMA (Time Division Multiple Access) such as GSM, also by the number ofassignable time slots per call specified in the Air Interface Standard.The total, or gross, bit rate comprises the source, or net, bit rate fortransmission of speech and the added bit rate derived from the redundantat least one bit per time slot used for channel error protection. Inpresent cellular systems, the ratio of the net bit rate and the extrabit rate originating from redundant bits is fixed.

There has generally been assumed that channels available fortransmission of speech have some more or less constant average quality.However, this is not always the case. Therefore, in existing systems, onone hand, when transmitting at a relatively high data rate, thetransmission can suffer from an insufficient channel error protection.On the other hand, when transmitting in other cases, the transmissioncannot be performed at an optimal data rate, due to a too strong errorprotection. A better or higher protection of a channel used transmissionfrom faulty transmitted information, i.e. a better channel quality,results in fewer bits being available for the transmission of theoriginal or source information, such as speech.

A method called Unequal Error Protection is used in most standards forcellular telecommunication systems. In Unequal Error Protection, bitscomprising speech information are divided into classes of decreasingperceptual importance and each class is encoded using an appropriaterate of protection. Although the Unequal Error Protection method used inmost standards to some extent mitigates the flaw of using a methodwherein a constant average value of the channel quality is assumed, thisis not the best possible solution.

The new Adaptive Multi-Rate (AMR) standard for the GSM system overcomesthe problem described above by being adaptive both with respect to thenet or source bit rate, i.e. the bit rate of the source information, andwith respect to the total or channel bit rate.

Adapting the coding rate of the source information is called codec modeadaptation and allows adapting the degree of error protection. At agiven fixed total bit rate, according to this method both the amount ofbits used for transmitting the source information and the amount ofredundancy bits that are added for protecting the channel from faultytransmitted bits are varied.

Adapting the gross or total bit rate is called mode adaptation herein.In a mobile telecommunication system working according to the GSMstandard two channel modes exist, the FR and the HR modes having totalbit rates of 22.8 and 11.4 kbps respectively.

The speech codec built according to the AMR specification is providedwith a number of codec modes having different bit rates, selected amongthe available source bit rates: 4.75; 5.15; 5.9; 6.7; 7.4; 7.95; 10.2;12.2 kbps. The amount of speech coding in relation to the amount ofchannel coding can be flexibly modified according to the requirementsset by the current channel conditions. Thus, in adapting the codec modeaccording to the AMR specification the present channel conditions aremeasured and the information obtained thereby is used to select thecodec mode that provides the optimum quality for the measuredconditions. Ideally, this allows achieving a speech quality curve of thecodec built according to the AMR specification that corresponds to theenvelope of the quality curves of the individual codec modes. FIG. 3 isa principle diagram of speech quality as a function of the channelquality for an exemplary speech codec built according to the AMRspecification having three modes. In this example, mode No. 3 has thehighest source bit rate and thus the highest speech quality undererror-free conditions. Modes Nos. 2 and 1 have lower source bit ratesand provide correspondingly non-optimal transmission under error-freeconditions. However, due to its relatively low error protection, codecmode No. 3 is sensitive to transmission errors and the transmission inthis mode deteriorates for channel conditions for which transmission inmode No. 2 and, particularly, in mode No. 1 still exhibits robustoperation. Transmission in codec mode 1 is most robust and can stilloperate under channel conditions in which transmission in the othermodes has already deteriorated.

Even if the speech codecs and the methods according to the AMRspecification were developed in order to improve the transmission ofspeech in the GSM system, due to its flexibility, the speech codecs andthe methods according to the AMR specification are also very suitablefor use in other systems. The speech codecs built according to the AMRspecification are used for the default speech service in UMTS/IMT-2000systems and are also very appropriate for voice-over-Internetapplications. In all systems, depending on the current network load, aspeech codec mode of a higher or lower rate may be selected.

AMR Adaptation

The principle of codec mode adaptation according to the AMRspecification will now be described. The source is an incomingelectrical signal representing speech information. The incoming speechsignal is channel encoded for the selected channel, using the currentlyselected codec mode and channel mode. The total resulting bits includingbits representing the speech information, bits for protection and bitsfor information regarding the selected mode, i.e. data for adapting thecodec mode, are together transmitted over the air interface. Data foradapting the codec mode consist either of channel measurement data, i.e.data indicating the estimated channel quality/capacity or a codec moderequest informing the sending side about the codec mode that the sendingside should select. Here, the sending side is defined as the encoder atthe mobile station or the encoder at the base station. The receivingside is defined as the decoder at the mobile station or the decoder atthe base station for the respective transmission case. A decoderreceiving a signal detects from the received signal the codec mode usedin the transmission and applies the decoding method of the detectedcodec mode to the received information representing speech. The receiveddata for adapting the codec mode, including information on the measuredchannel conditions or a codec mode request, is used for selecting thecodec mode for the next signal transmitted from the encoder at thereceiving side. Moreover, the decoder at the receiving side also makesmeasurements on the received signal, which result in new channelmeasurement data, i.e. data representing the current channel conditionsin receiving.

Channel measurement data can—after suitable quantization—be directlytransmitted to the sending side or it can first be fed into an adapter,i.e. a selector of the codec mode of the incoming channel. The adaptergenerates a codec mode request or a command in response to themeasurement data, which is an indication of the codec mode to be used bythe sending side. When this adapter is located on the receiving side,the corresponding codec mode request/command is sent to the sending sideinstead of the original measurement data. However, when the adapter islocated at the sending side, signals representing the measurement datahave to be transmitted to sending side. A binding codec mode request isusually referred to as a “codec mode command” whereas if it is merely anindication of the preferred mode and the sending side has the authorityto override it, it is referred to as a “codec mode request”. Thisdistinction is of minor relevance herein. In the following the acronymCMR will be used for both “codec mode command” and “codec mode request”.

CMRs are generated by the adapter based on a measurement of the channelquality. A mapping operation of the measured data to the CMRs isperformed. The mapping operation may involve the comparison of themeasurement values to predetermined threshold levels. Usually ahysteresis is implemented in this mapping, i.e. different thresholdlevels are used depending on if—with respect to the previous CMR—a CMRcorresponding to a more robust or a less robust mode is to be generated.

A measurement value of the channel quality can be calculated frommeasurements, e.g. burst-wise measurements of the signal-to-interferenceratio (C/I), in the case of a radio channel, or estimates of the totalbit error rate. The calculation usually involves filtering instantaneousmeasurement values by a filter having a memory since measurements takenfrom only one burst or one frame fluctuate too strongly. In a GSMsystem, a burst of data is transmitted approximately every fifthmillisecond. A frame is a unit within which speech and channel coding isperformed. The purpose of filtering is to generate a measurement valuethat deviates less from the true value than the measurements of a singleburst or frame, respectively. Typical filters are linear smoothing andprediction filters. Examples of such filters are given in the GSMpublication “GSM 05.09: Link Adaptation”.

The performance of link adaptation according to the AMR specification ismanifested by the capability of the system to adapt the used codec modeto the present channel condition. That is, a system having optimalperformance when applying link adaptation according to the AMRspecification will always use a codec mode that gives the best possiblesignal quality at the present channel condition, whereas situations inwhich an unsuitable mode is selected do not occur. Unfortunately, thisideal state cannot be reached by practical systems. However, it iscrucial to avoid situations in which modes not providing sufficientchannel error protection for the present channel condition are selected.Such situations cause extreme signal quality degradation resulting in atotal deterioration of the channel error protection. The selection of amode that is too robust is not so critical as it does not result in adrastic quality drop. This can be tolerated at least for limited periodsof time.

A system parameter in a system using the AMR specification being crucialfor the performance of link adaptation according to the AMRspecification is the AMR loop delay. The AMR loop delay is the timerequired to transmit a new codec mode command until a signal, encoded inthe new codec mode, is received at the decoder end of the channel. TheAMR loop delay consists of several components:

-   Delay at the measurement filter.-   Delay due to transmission of the measurement signals or CMR back to    the source coding device.-   Delay due to source coding.-   Delay due to channel coding and the scrambling of the channel    encoded bits in their time position.-   Delay due to the time it takes to perform signal processing-   Delay due to buffering and frame alignment.

The total loop delay in systems designed according to the AMRspecification varies from system to system and depends on severalfactors such as:

-   1. the type of link between the encoder at the sending side and the    decoder at the receiving side, e.g. radio or wire.-   2. the type of connection between the channel codec in the base    station and the speech codec (TRAU, Transcoding and Rate Adaptation    Unit) in the BSC or MSC, e.g. wire, microwave, satellite channel.-   3. the type of connection from one end to the other end, e.g. MS    (mobile station) to PSTN (Public Switched Telephone Network), or MS    to MS using tandem encoding or tandem-free-operation.    Tandem-free-operation (TFO) is a transmission mode used for the    speech codec data, in which no speech coding using AMR is employed    at the base station or a mobile switching center (MSC) or Radio    Network Controller (RNC). Hence, the bits coding speech generated by    the speech encoder of the first end of a communication channel are    directly propagated to the other end of the channel that performs    the decoding. TFO can be used in MS to MS connections but also e.g.    in connections to or from a codec using AMR of an IP client.    Connections with TFO have a relatively high AMR loop delay. MS to MS    connections with TFO have two communication channels so that when    applying AMR codec mode adaptation a codec mode suitable for both    communication channels must be selected.-   4. whether the channel is an uplink or downlink,-   5. whether discontinuous transmission is used or not,-   6. the switching method of the connection, i.e. whether it is    circuit-switched or packet-switched (Internet Protocol, Asynchronous    Transfer Mode).

Depending on these factors the AMR loop delay may vary from about 100 msup to more than 500 ms.

Concepts of Systems According to the AMR Specification

Different kinds of AMR-system are conceivable, e.g. those havingdistributed, centralized and mixed control of codec mode adaptation.Systems having distributed control are symmetric between base stationand mobile station. Each receiving side, the decoder at the mobilestation and the decoder at the base station, performs qualitymeasurements on the transmission in the incoming channel and sends linkadaptation commands or measurement data to the transmitting side.

In systems having centralized control, the adaptation control of boththe uplink and the downlink resides at the base station. The mobilestation merely assists the network in the adaptation of the downlink byconveying channel quality measurements to the network. There the CMR forthe downlink is generated and it is binding. There is no difference ofthe uplink adaptation between systems having distributed and centralizedcontrol.

Mixed control is a combination of centralized and distributed controlconcepts. It is specified in the AMR specification for the GSM system. Asystem having mixed control has distributed control as described but thebase station can override the CMR from the mobile station with which thebase station communicates.

Problem

If the physical conditions of the transmission in a channel vary slowlycompared to the AMR loop delay, then good performance of AMR linkadaptation can be easily achieved if, when performing an AMR linkadaptation, the mode being most appropriate for the presently measuredchannel condition is selected. If, on the other hand, there are veryfast fluctuations of the channel conditions, it is possible to apply asuitably designed low pass filter. Such a filter can be capable ofremoving the fast fluctuations from the measurements. Thus, the filtergenerates a measurement value corresponding to the mean value of thefluctuations.

The link adaptation according to the AMR specification described in theGSM specification GSM 05.09 “Link Adaptation” follows this principle. Itapplies linear low-pass filtering to burst-wise generated C/I estimates.No severe problems have been observed when using this kind ofadaptation. However, so far, only system configurations having low AMRloop delays have been considered.

However, if the AMR loop delay is as high as, e.g. 500 ms which may bethe case for TFO connections, or in networks having satellite linksbetween BTS (Base Transceiver Station) and TRAU (Transcoding and RateAdaptation Unit), the use of the existing adaptation method according tothe AMR specification may result in problems. So-called out of phasechannel conditions may occur, which are caused by the delay between afirst event involving a changed channel condition and a second eventinvolving the start of using a codec mode suitable for the alteredchannel condition. Specifically, link adaptation according to the AMRspecification may require a certain mode based on the present channelcondition, but due to the AMR loop delay, when that mode starts to beapplied, the channel conditions may have changed drastically. Hence, thechosen mode is no longer optimal for the actual channel condition, i.e.an out of phase condition has occurred. This is particularly a problemif a less robust mode than the one currently used has been requested,and when the mode is finally applied, the quality of the currently usedchannel has deteriorated. This is likely to cause frame erasures and anincreased bit error rate and ultimately results in a bad quality of thereconstructed speech signal.

The problem is exemplified by the case where the channel conditions varyperiodically with a period of about twice the AMR loop delay. At thiscritical fluctuation rate of the channel condition, the adaptation willbe maximally out of phase. A robust mode is used when the channel isgood and a less robust mode is used when the channel is bad. In FIGS. 2a and 2 b, the problem is illustrated by an artificial channel profile,which was generated using a channel simulator having a predeterminedaverage value of the channel quality. The channel quality follows aswept sinus curve, i.e. a sine wave having an increasingly higherfrequency. The codec mode is out of phase with the actual channelcondition at a certain channel fluctuation frequency, as can be seen inthe figures. It can be seen that inappropriate codec modes are selectedand that a high number of frame erasures (fe) occur. The lower curves inFIGS. 2 a and 2 b show the varying channel quality having an averagecarrier-to-noise (C/N) ratio between 3 dB and 15 dB. The upper curvesshow the selected codec mode, which in this example can be the 12.2,7.95 or 5.9 kbps mode. Frame erasures (fe) are indicated by the sign“x”. The total number of frame erasures is in this example 31.

The problem described above has not previously been encountered ordiscussed since no situations with high AMR loop delays have beenassumed. Thus, it was possible to apply low-pass filtering of thechannel measurements, according to the GSM specification GSM 05.09. Thisfilter has a cut-off frequency below the critical channel fluctuationrate at which frequency the adaptation becomes out of phase. Thus, thisfilter attenuates at the critical fluctuation rate. Therefore, thefilter removes fluctuations, at least to a large extent. This means thatthe filter evens the fluctuations and generates an output valuecorresponding to the mean value of the incoming fluctuations. Such afiltered measurement value is more suitable for AMR link adaptation.

For high AMR loop delays, a similar method might appear obvious. A lowpass filter would be required having a cut-off frequency below a therate corresponding to (½*AMR loop delay). Requiring the filter to havesuch a cut-off frequency obviously introduces an additional filterdelay, which increases the total AMR loop delay even more andconsequently does not solve the problem. This is a consequence of theuncertainty principle, which is a fundamental physical law, causing thefilter delay to be inversely proportional to the filter bandwidth.

Prior Art

In the published European patent application No. 0 964 540 for KrinasamyAnandakumar et al. and assigned to Texas Instrument Inc., a system isdisclosed for dynamic adaptation of data channel coding in wirelesscommunication between a mobile station and a base station. A frametransmitted from the mobile station includes a convolutionally codedportion containing a downlink measurement bit and a repetition codeidentifying the codec mode of the frame. A frame transmitted from thebase station includes a codec mode command for the mobile station in theconvolutionally encoded portion, and a repetition code identifying thecodec mode of the downlink frame. The base station includes means foranalyzing the quality of the uplink frame and means for determining thequality of downlink from the received downlink measurement. The systemis designed as one possible implementation of an AMR-system for GSM. Thedeficiency of the system and method disclosed in this European patentapplication is that it does not consider a situation with a high AMRloop delay.

In U.S. Pat. No. 5,701,294 for Ward et al. and assigned to TexasInstruments Inc., a system and a method are disclosed for flexiblecoding in a radio communication network. The system continuouslymonitors radio channel quality in both uplink and downlink transmission,and dynamically adapts the combination of speech coding, channel coding,modulation, and the amount of assignable time slots per call to optimizethe quality of the transmitted signal at the current channel conditions.Various combinations of the speech coding, channel coding, modulation,and assignable time slots are identified as combination types andcorresponding cost functions are defined. However, the system and methodproposed in this U.S. patent are not adapted to the new GSM AMR systemsas specified by ETSI. Furthermore, the problems related to a high AMRloop delay are not discussed.

The article by Erdal Paksoy et al., “An Adaptive Multi-Rate Speech Coderfor Digital Cellular Telephony”, 1999 IEEE International Conference onAcoustics, Speech, and Signal Processing, Proceedings, 1999, Vol. 1, pp.193-196, discloses an adaptive multi-rate (AMR) speech coder designedfor digital full rate (22.8 kb/s) and half rate (11.4 kb/s) channelsaccording to the GSM specifications used to maintain a high quality inthe presence of highly varying background noise and channel conditions.The decoders monitor channel quality at both ends of the wireless linkusing the soft values for the received bits and assist the base stationin selecting the codec mode appropriate for a given channel condition.The adaptation algorithm used for the channel measurement comprisesestimating the carrier to interference ratio (C/I). This estimate isbased on the soft values, i.e. values comprising reliabilityinformation, for the received bits as provided by thedemodulator/equalizer. These values are good indicators of thereliability of the bits. The IEEE article states that a moving averageof the absolute values of the soft bits is a good estimator of thecurrent signal-to-interference ratio of the channel. Codec modedecisions are made by comparing the moving average to a predeterminedthreshold, and by using additional hysteresis rules designed to ensuresmoother transitions when changing to a new codec mode. Because of theirdifferent characteristics, the full rate and half rate channels requirethe various parameters of the adaptation mechanism to be tunedseparately. However, the problems related to a high AMR loop delay arenot discussed. In particular, in this article, filtering using a movingaverage is suggested, which is a kind of linear low pass filtering. Asoutlined in the problem discussion above, conventional low passfiltering provides no solution to the problem.

In the published European patent application No. 0 986 206 for PatrickCharriere and assigned to Alcatel a method of changing the encodinglevel of digital data transmitted between a transmitter and a receiverat a constant rate is disclosed. The encoding level is determined as afunction of the signal received by a receiver. The change of theencoding level to a less robust one is delayed by fixed time in order toensure that the encoding level chosen will not result in too many lostframes.

SUMMARY

It is an object of the invention to provide a method for efficientadaptive mode selecting.

It is another object of the invention to provide a digital communicationsystem for efficient adaptive mode selecting and a delay unit for adigital communication system.

The main problem associated with existing adaptive multi rate systems isthat due to a delay between the measurement operation and the adaptationoperation, especially when the delay is high, the adaptation isperformed at an unsuitable time.

The solution to the problem described above is to apply a suitablyselected filtering operation in order to organize the values to befiltered one after the other. At each time instant, the sample values inthe memory are arranged in an order according to some algorithm and onesample having a predefined time index is output. A filter performingthis kind of filtering on channel measurements may be located prior toan adapter. However, said filtering operation may also be applied toCMRs being the output of an adaptation device. As an example, the filterdelays CMRs indicating or requesting a less robust codec mode, whereasCMRs for a more robust codec mode are propagated without any delay. As aconsequence, a more cautious behavior of AMR adaptation is achieved.Switching to a less robust and thus more risky mode is performed onlyafter a delay, when there is an increased likelihood that the channelconditions remain good.

It is an object of the invention to provide a method for codec modeadaptation in which selection of such codec modes is avoided, which arenot sufficiently robust to allow a secure transmission for the currentchannel condition.

An advantage resulting from using such method is that in thetransmission there will be a decreased number of frame erasures and anincrease of the overall speech quality when finally played to alistener. AMR systems using the proposed method will generally be morerobust.

The proposed method generally results in a more optimized selection ofcodec mode. If the channel is really improving, a less robust modehaving a higher intrinsic quality is selected only after a delay. Thismay reduce the quality of the received speech signal to some extent.However, the potential quality loss is clearly compensated by the factthat the method as disclosed herein provides a way of minimizing the outof phase problems due to AMR loop delay. It should be observed that thedegradation of the speech quality could be significant if a notsufficiently robust mode is used, whereas the speech quality degradationcaused by keeping a codec mode being more robust than necessary for atoo long time is very moderate.

The term “comprises/comprising” when used in this specification is takento specify the presence of stated features, integers, steps orcomponents but does not preclude the presence or addition of one or moreother features, integers, steps, components or groups thereof.

Further scope of applicability of the present invention will becomeapparent from the detailed description given hereinafter. However, itshould be understood that the detailed description and specificexamples, while indicating preferred embodiments of the invention, aregiven by way of illustration only, since various changes andmodifications within the spirit and scope of the invention will becomeapparent to those skilled in the art from this detailed description.

Additional objects and advantages of the invention will be set forth inthe description which follows, and in part will be obvious from thedescription, or may be learned by practice of the invention. The objectsand advantages of the invention may be realized and obtained by means ofthe methods, processes, instrumentalities and combinations particularlypointed out in the appended claims.

BRIEF DESCRIPTION OF THE DRAWINGS

While the novel features of the invention are set forth withparticularly in the appended claims, a complete understanding of theinvention, both as to organization and content, and of the above andother features thereof may be gained from and the invention will bebetter appreciated from a consideration of the following detaileddescription of non-limiting embodiments presented hereinbelow withreference to the accompanying drawings, in which:

FIG. 1 is a block diagram illustrating a telecommunication system inwhich the AMR coding scheme is used when the signal is out of phase, innon-tandem free operation,

FIGS. 2 a and 2 b are diagrams illustrating codec mode decision inpresent AMR systems,

FIG. 3 is a diagram illustrating the principle of AMR codec modeadaptation,

FIGS. 4 a and 4 b are diagrams illustrating code mode decision for anAMR system having delayed codec mode switching,

FIG. 5 is a block diagram illustrating an alternative analyzer,

FIG. 6 is a block diagram of a delay unit, and

FIG. 7 is a block diagram illustrating a telecommunication system inwhich tandem free operation is used.

DETAILED DESCRIPTION

A telecommunication system will be described operating at least in partin a radio or wireless communication environment. However, the methodsand devices to be described hereinafter can be used in any digitalnetwork, using radio and/or wired connections and using some kind ofencoding of digital voice or speech information, the not encoded ordecoded voice or speech information directly representing acousticsignals in the conventional way, as recorded by a microphone.

The block diagram of FIG. 1 illustrates a mobile, wirelesstelecommunication system comprising a base station 200 and a mobilestation 100 communicating with each other on an uplink channel 305 and adownlink channel 315. In both the base station and the mobile stationAMR coding schemes are used. The mobile station 100 comprises aspeech/channel encoder 110 that can adopt different modes of coding ordifferent coding schemes having different degrees of robustness, ananalyzer 120 for sensing and analyzing the condition of the downlinkchannel, a selector 130 of downlink mode, and a speech/channel decoder160. The base station 200 comprises a speech/channel encoder 260 thatlike the encoder 110 of the mobile station can adopt different modes ofcoding or different coding schemes having different degrees ofrobustness, an analyzer 220 for sensing and analyzing the condition ofthe uplink channel, a selector 230 of uplink mode, and a speech/channeldecoder 210. The base station may also comprise a selector 240 ofdownlink mode, either instead of or in addition to the selector 130 inthe mobile station. In the latter case the selector 240 of downlink modehas the authority to override CMRs generated by the selector 130 ofdownlink mode located in the mobile station. The mobile station 100 mayalso comprise a selector 140 of uplink mode. The mobile stationtransmits on the uplink channel 305 information which has been encodedby its encoder 110 and which is decoded by the decoder 210. The basestation 200 transmits information on the downlink channel 315 which hasbeen encoded by its encoder 260 and which is decoded by the decoder 160.It should be noted that the speech decoder 210 and the speech encoder260 in the base station do not have to be physical parts of orphysically located in the base station. The encoder and the decoder canbe located in an MSC, not shown, connected to the base station 200.

In operation, when communicating information representing speech, thedecoder 160 in the mobile station 100 receives a signal transmitted onthe downlink channel 315 from the base station 200. Then, the decoder160 decodes the received signal to produce speech signals, see the arrow165, that are made audible to the user, not shown, of the mobilestation. Furthermore, the decoder 160 also decodes or detects, in thereceived signal, codec mode information derived from and/orindicating/comprising the measured quality or condition of the uplinkchannel 305. This information can be or include a codec mode command orrequest for uplink transmission that is then fed to the encoder 110 inthe mobile station to set the encoder to work in the mode indicated inthe command. In the case where the information does not directlyindicate a codec mode the codec mode information is fed to the optionalselector 140 in the mobile station, in which a new mode for uplinktransmission is determined or selected. Then, the result from theselector 140 is fed to the encoder 110 to set it to work in the modeindicated in the result so that this mode is used in the next uplinktransmission. Incoming speech signals, see the arrow 115, from the userof the mobile station are speech and channel encoded in the encoder 110of the mobile station according to the selected or set codec mode.Thereafter, encoded speech is transmitted via the uplink channel 305 tothe base station 200.

Simultaneously, the signal received in the mobile station 100 from thebase station 200 is fed to the analyzer 120 in the mobile station inwhich the signal is sensed and analyzed to determine the communicationcondition or the quality of the communication on the downlink channel315. Then, the result of the analysis is fed to the selector 130 ofdownlink mode in the mobile station in which, based on the result of theanalysis, a codec mode command or request for downlink transmission isgenerated. A man skilled in the art understands that very often thecodec mode used for downlink transmission has to be changed depending onthe channel state. Then, the generated codec mode command or request isfed to the encoder 110, which encodes the codec mode request to betransmitted to the base station 200. In the case where selector 130 isprovided in the mobile station, the result of the analysis in theanalyzer 120 is fed directly to the encoder 110.

The decoder 210 of the base station 200 receives a signal from themobile station 110 on the uplink channel 305. The decoder decodes thereceived signal and transforms the relevant part of it to speechsignals, see the arrow 215, which are transmitted to the party withwhich the user of the mobile station is talking. Furthermore, thereceived signal is also fed to the analyzer 220 of the base station inwhich the signal is analyzed to determine the quality of thecommunication on the uplink channel 305. Then, the result from theanalyzer 220 is fed to the selector 230 connected to or in the basestation in which a codec mode is selected to be used for uplinktransmission. The result from the selector 230 is fed to the encoder 260of the base station to be transmitted to the mobile station 100 on thedownlink channel 315. Alternatively, the result from the analyzer 220 ofthe uplink channel is fed directly to the encoder 260 to be transmittedto the mobile station 100.

Furthermore, the decoder 210 in the base station 200 decodes or detectsin the received signal codec mode adaptation data such as a codec moderequest for the downlink channel or, alternatively, data representingthe analyzed condition of the communication on the downlink channel. Inthe case where the data comprises a codec mode command it is feddirectly to the encoder 260 to set the encoding mode thereof, theencoder encoding speech incoming from the other party of theconversation, see the arrow 265. In the alternative case, the decoded ordetected data from the decoder 210, these data comprising either a codecmode request or measurement data, are provided to the optional selector240 in which a codec mode is selected. Thereafter, the result comprisingthe selected mode is fed to the encoder 260 to set it to work in thatmode.

Thus generally, codec mode adaptation requires the transmission ofadaptation data for the considered transmission channel or link. On theuplink channel 305, adaptation data for adapting the downlinkcommunication are transmitted. On the downlink channel 315, adaptationdata for adapting communication on the uplink are transmitted. Theadaptation data comprises CMRs or channel measurement data, the latterbeing values obtained or derived in the analysis performed by theanalyzers 120 and 220 respectively of the received signal andrepresenting the communication condition of the respective channel.

Now, the adaptation for uplink transmission will be described. The basestation 200 monitors the condition of the uplink channel 305 and decidesthe codec mode that the mobile station 100 should use. Therefore, theanalyzer 220 analyzes the signal received from the uplink channelthereby determining or measuring the quality of the communication in thechannel. The result of the analysis is sent from the analyzer 220 to theselector 230 of uplink mode in the base station. The selector 230selects an uplink codec mode suitable for the current condition of theuplink channel. The base station 200 communicates this information as aCMR, transmitted on the downlink channel 315 to the mobile station 100.Upon reception, the encoder 110 in the mobile station switches to themode indicated in the CMR. In the alternative, the mobile station 100accommodates an alternative selector 140 of uplink mode, instead of theselector 230 at the base station. In that case values representing thedetermined condition of the uplink channel are output from the analyzer220 of the uplink channel and are transmitted via the downlink channel315 to the alternative selector 140 of uplink mode.

Now, link adaptation for downlink transmission will be described. Basedon the signal received from the base station 200 over the downlinkchannel 315 and possibly other information that may be available, theanalyzer 120 in the mobile station 100 determines or measures thecondition of the communication in the downlink channel to find thequality of the communication of the channel. The result that is producedby the analyzer and can comprise values representing the determinedquality is fed to the selector 130 of downlink mode in the mobilestation which generates a CMR based on its input. The generated CMR istransmitted via the uplink channel 305 to the base station 200 in whichit is received and can be directly transferred to the encoder 260 forsetting the encoding mode. As an alternative, the CMR received in thebase station may be fed into the optional additional selector 240 ofdownlink mode that has authority to override the received CMR. Theoutput of the optional selector 240 of down link mode indicates theselected codec mode to the encoder 260 of the base station. In the casewhere the mobile station does not accommodate a selector of downlinkmode, values or measurement data being the result of the analysis of thecommunication in the downlink channel and representing the quality ofsaid communication are sent via the uplink channel 305 to the basestation 200. There, the received values are fed to the optional oralternative selector 240 of downlink mode.

In FIG. 7 a block diagram of a system is shown having established atandem-free-operation (TFO) connection between a mobile station or an IPclient 700 and a mobile station or an IP client 710, respectively, via afirst base station 750, a transport network 770 and a second basestation 780. As the configuration is symmetric for transmission ofinformation representing speech in both directions, the system will bedescribed only for speech transmission from the mobile station or the IPclient 700 to the mobile station 710. An encoder 701 receives a speechsignal on a line 703 and encodes it according to its current mode ofcodec operation, as has been set or indicated by a received controlsignal on the control line 704, and also makes a channel encoding. Thecoded speech signal is transmitted via an uplink channel 731 to thefirst base station 750. A channel decoder 751 in the first base stationchannel decodes the received data, i.e. the coded speech signal, andforwards the channel decoded data through the transport network 770 to achannel encoder 781 in the second base station 780, in which the data ischannel encoded for transmission over a downlink channel 732 to theother mobile station or IP client 710. The first base station 750further comprises an analyzer 760 of uplink channel that performsmeasurements on the signals received over uplink channel 731 todetermine the condition of the communication in said channel and to findthe quality thereof. Values or data representing the result of theanalysis of the communication in the uplink channel are provided to aselector 761 of uplink codec mode, also in the first base station 750,which generates preliminary codec mode command data CMR1 that are thenfed to a combiner 762 in the first base station.

The mobile station or IP client 710 comprises a channel and speechdecoder 711 which decodes coded speech data incoming from the downlinkchannel 732 and outputs the recon-structed speech signal on an outputline 715. The mobile station or IP client 710 further comprises ananalyzer 712 that performs measurements on the signals received from thedownlink channel 732 to determine the condition of the communication inthe channel and the quality of the communication. Then, values or dataresulting from the channel analysis and representing the condition orquality of the communication in the downlink channel are fed from theanalyzer 712 in the mobile station or IP client 710 to a selector 713 ofdownlink codec mode, which generates preliminary codec mode command dataCMR2. These data CMR2 are channel encoded in an encoder 714 in themobile station or IP client 710 and are then transmitted, together withinformation representing speech, via an uplink channel 733 to the secondbase station 780. There, a channel decoder 782 decodes received signalsand finds the CMR2 and provides it via the transport network 770 to thecombiner 762 in the first base station 750. Optionally, the CMR2 isforwarded to the combiner 762 via an additional selector 763 of codecmode in the first base station that has the authority to override CMR2to provide a modified CMR2. The optional selector 763 of codec mode isalso used in the case where the selector 713 of codec mode in the mobilestation or IP client is not used, and the data representing the qualityof the channel is then instead provided from the analyzer 712 ofdownlink channel in the mobile station or IP client 710 to the optionalselector 763 in the first base station 750 via the encoder 714, theuplink channel 733 and the decoder 782. The optional selector thenproduces the CMR2. The combiner 762 combines the preliminary codec modecommands or requests CMR1 and CMR2 to a final CMR3 that is output fromthe combiner 762. Then, the final CMR3 is fed to a channel encoder 754in the first base station 750, in which the CMR3 is channel encoded andis then transmitted over a downlink channel 734 to the first mobilestation or IP client 700. There, a channel decoder 702 decodes thesignals received on the downlink channel to find inter alia, CMR3 andthen provides this command to the encoder 701, to set the codec mode ofthe encoder. The channel encoder 781 is arranged to use the same codecmode as the encoder 701 for the same transmitted information, the codecmode used being decoded or detected by the decoder 751 in the first basestation 750.

In FIG. 6 a block diagram of a delay unit for codec mode commands orrequests, CMRs, is shown. Each time when a CMR is available to set theoperative mode of an encoder from a selector of codec mode, as has beendescribed above, it is input to the delay unit. The delay unit generatesa CMR on its output as a response to the current input CMR and at leastone previously input CMR. The delay unit is provided with a memory 810in which a number M of the most recent, received CMRs are stored, Mbeing an integer and designating the memory length of the delay unit, asuitable choice being e.g. M=25. A selector 820 is connected to thememory locations X₁, X₂, . . . , X_(M) of the memory 810. The selector820 controls a switch 830. By the switch the content of a selectedmemory location of the memory 810 is passed to the output of the delayunit. The selector 820 controls the switch 830 according to a filteralgorithm 825.

According to a suitable filter algorithm that memory location isselected which contains that CMR among the CMR stored in the memory 810that is indicative of the most robust codes mode. Such a process can befacilitated by e.g. assigning integer numbers to the codec modes, wherethe smaller the integer number is, the more robust is the codec mode,e.g. the most robust mode MR475 is assigned the integer value 1, thesecond most robust mode MR515 is assigned the integer value 2, . . . ,and the least robust mode MR122 is assigned the integer value 8. Then,the memory location holding a CMR that has the lowest integer numberamong those associated with the stored CMRs is selected.

The filter algorithm may also search for that CMR among the CMRs storedin the memory 810 which indicates the n-th most robust codec mode, wheree.g. n=2. The filter outputs the selected CMR. The advantage ofselecting the n-th most codec mode is that a trade-off between qualityand the speed of the codec mode adaptation can be performed according tothe current channel conditions.

As an alternative to the filter algorithm disclosed above, the filteralgorithm calculates and outputs the median of the CMRs stored in thememory 810. In this case the CMRs stored in the memory are sorted withrespect to the robustness of the corresponding codec mode and generallythe m-th element (1≦m≦M, m being an integer) of the sorted CMRs is usedas output of the delay unit. For providing the median, the delay unitoutputs the middle element for which m=floor((M+1)/2).

According to another embodiment the filter operation according to thedifferent, alternative algorithms as described in conjunction with FIG.6 are applied to values representing the current condition or quality ofthe channel before or after possible linear filtering. Linear filteringis usually part of the analysis devices 220, 120, 760, 712, but thelinear filtering may also be part of the selectors 130, 140, 761, 713,763.

The last example of the filter algorithm performed by the delay unit inFIG. 6 involves the step of first rearranging the order of the data inthe cells of the memory 810 to place the stored data in somepredetermined way or according to some algorithm and then selecting thedata of a predetermined memory cell or memory location as output.Alternatively, this can be interpreted as selecting the value of amemory location X_(i), the index i being an integer in the range of 1 toM, in the memory 810, the choice of the index i being dependent on thecontents of the memory. In the examples discussed above, it is possibleto achieve an adaptation behavior, which allows different adaptationspeeds depending on whether switching is performed with respect to thepresent codec mode towards a more robust mode or a less robust mode. Inparticular this allows immediate switching to more robust modes whereasswitching to less robust modes is delayed until it is rather probablethat the channel condition remains good enough for that less robustmode, as will be described below.

The method described above is illustrated by the following example.Assume that the memory length of the delay unit is M=5, and that the AMRspeech codec operates at two codec modes, MR475 and MR122, the modeMR475 being more robust than mode MR122. The filter algorithm isdesigned so that the delay unit returns a CMR selected among the CMRs inthe memory and corresponding to the most robust mode. First, it isassumed that the channel has a good quality. Thus, all memory locationscontain CMRs for mode MR122. The delay unit outputs MR122. A suddendegradation of the channel quality leads to a CMR of the more robustmode MR475 at the input of the delay unit. Then, the memory of the delayunit comprises the mode MR475 in location X₁, and the mode MR122 inlocations X₂, X₃, X₄, X₅. According to the filter algorithm, the delayunit immediately outputs MR475, i.e. the adaptation immediately reactson channel quality degradations. Now, the opposite case is assumed, i.e.that the communication in the channel is bad. Then, all memory locationscomprise CMRs for the mode MR475. A sudden improvement of the channelresults in a CMR for mode MR122 at the input of the delay unit. However,the delay unit still generates a CMR for mode MR475 as there are stillmemory locations in the delay unit comprising the mode MR475. It takessome time before all the five memory locations comprise CMRs for theless robust mode MR122 and thus before operation at the less robust modeMR122 is requested. Hence, the channel must have a good quality for apredefined time period, since a single CMR for a more robust mode in thememory of the delay unit does not trigger any change of mode. Thus, theadaptation to a less robust mode is delayed until it is more certainthat the channel has an improved quality.

The delay unit is of major importance in the method that is describedherein and the principle of the method cannot be implemented without thedelay unit. The length M, i.e. the number of memory locations, of thememory of the delay unit is a design parameter influencing the behaviorof the AMR adaptation. According to a preferred embodiment thisparameter is selected adaptively with respect to the AMR loop delay. Inparticular, the memory length M of the delay unit can be proportional tothe AMR loop delay. As an example, the length of time corresponding tothe memory length M can be equal to the AMR loop delay. For instance,assuming a GSM system, the CMRs are generated at a rate of one every 40ms. If the AMR loop delay is 480 ms, then the length M is chosen to480/40=12. If, however, the AMR loop delay is only 200 ms, the memorylength M is chosen to be equal to 5.

As an option, additional filtering, i.e. an adaptive delay operation,described herein may only be performed when the AMR loop delay is higherthan a threshold d, e.g. the threshold d being equal to 200 ms.

The method described herein can be implemented in various systemconfigurations. According to a preferred embodiment the additionalfiltering is performed in the analyzers of the respective incomingchannel, i.e. the analyzer 120 in the mobile station 100 in FIG. 1 inthe case of the downlink channel and the analyzer 220 in the basestation 200 for the uplink channel. In this case, the additionalfiltering is performed using the measurement values and not CMRs.

In FIG. 5 the structure of an alternative analyzer 1000 is shown. Theanalyzer 1000 comprises three functional blocks, a primary analyzer 910,a delay unit 920 and a linear filter 930. First, the incoming signal istransmitted to the primary analyzer 910, which has the same function asthe analyzers 120, 220 described above. Then, the signal is transmittedto the delay unit 920, which has the same function as the delay unitdescribed above. Thereafter, the signal is fed to the linear filter 930.Linear filtering of measurement values may be part of the originalanalyzer 120, 220. In the analyzer 1000, the linear filtering ofmeasurement values is provided after the delay unit 920, in the linearfilter 930.

According to another preferred embodiment of the proposed method theadditional filtering is performed in the selectors located at therespective receiving ends. For downlink adaptation the additionalfiltering is then performed in the selector 130, and for uplinkadaptation in the selector 230, see FIG. 1.

According to a further preferred embodiment of the method the additionalfiltering is performed in the selectors located at the respectivetransmitting ends. For downlink adaptation, this is then performed inthe alternative selector 240, and for uplink adaptation the additionalfiltering is performed in the alternative selector 140, see FIG. 1.

The method described herein can also be advantageously used for TFOconnections. The additional filtering may be performed in the mobilestation/IP client 710 either as part of the analyzer 712 of the downlinkchannel or as part of the selector 713 of codec mode for downlinktransmission, see FIG. 7.

In a further example, the additional filtering is performed in part bothin the additional selector 763 of downlink codec mode and in part eitherin the analyzer 760 of uplink channel or in the selector 761 of uplinkchannel, respectively, in the relevant base station, see FIG. 7. Datarepresenting the condition of the communication in the downlink channelis fed to the selector 763 in which additional filtering is performed.It should be noted that for the downlink channel 732 the AMR loop delayis larger than the AMR loop delay for the uplink channel 731, since AMRadaptation for the downlink involves a big control loop starting fromthe second mobile station/IP client 710 via the uplink 733, the secondbase station 780, the transport network 770, the first base station 750,the downlink 734, the first mobile station or IP client 700 and back tothe second mobile station or IP client 710 via the uplink 731, the firstbase station 750, the transport network 770, the second base station 780and the downlink 732. The additional filtering performed in the analyzer760 or the selector 761 is performed using data representing the stateof the uplink channel 731 for which the AMR loop delay is smaller asonly a small or short control loop is involved, this loop including thefirst base station 750, the downlink 734, the first mobile station or IPclient 700 and the uplink 731. Thus, it is more advisable to applydifferent memory lengths M for the different delay units, correspondingto the different AMR loop delays. The combiner 762 selects that CMR3among CMR1 and CMR2, which corresponds to the most robust codec mode, asthe output signal.

Furthermore, in the system of FIG. 7, it is possible to perform theadditional filtering after the combiner 762 in the relevant base stationor after decoding in the decoder 702 in the mobile station or IP client700. However, this is less advantageous as it does not provide thebenefit of using different memory lengths M for the different effectiveAMR loop delays.

FIGS. 4 a and 4 b show channel profiles for the case when the method asdescribed herein is used. The curves in FIGS. 4 a and 4 b have beengenerated using a channel simulator having a predetermined average valueof the channel quality. The channel quality follows a swept sinus curve.As is shown in FIGS. 4 a and 4 b, out-of-phase conditions betweenchannel quality and selection of codec mode are avoided, when the methoddescribed above is used. Codec modes not being sufficiently robust arenot selected anymore. The advantage of the method is also visible bycomparing the number of frame erasures, which have been reduced from 31to 3. By using the method described herein the perceived quality of thereceived signal is significantly improved.

While specific embodiments of the invention have been illustrated anddescribed herein, it is realized that numerous additional advantages,modifications and changes will readily occur to those skilled in theart. Therefore, the invention in its broader aspects is not limited tothe specific details, representative devices and illustrated examplesshown and described herein. Accordingly, various modifications may bemade without departing from the spirit or scope of the general inventiveconcept as defined by the appended claims and their equivalents. It istherefore to be understood that the appended claims are intended tocover all such modifications and changes as fall within a true spiritand scope of the invention.

1. A method of controlling the operation of an Adaptive Multi-Rateencoder for encoding speech signals to be sent between a first node anda second node in a wireless communication system, the encoder beingoperable to encode the speech signals in at least two modes havingdifferent degrees of robustness, the method comprising the steps of:sensing a first communication condition in a communication channelbetween the first and the second node; identifying a first modeapplicable to said encoder corresponding to the first communicationcondition in the communication channel; storing a value indicative ofthe first mode in a memory, the length M of which is selected responsiveto an Adaptive Multi-Rate loop delay, where M is an integer; andcomparing the value indicative of the first mode to a plurality ofvalues previously stored in the memory, each of said plurality of valuesbeing indicative of one of a set of previous modes corresponding to oneof a plurality of previously sensed communication conditions wherein,compared to the plurality of previously stored values, if the valueindicative of the first mode indicates that the first communicationcondition is degraded relative to the plurality of previously sensedcommunication conditions, selecting said first mode as a selected modeto be applied by said encoder; or is improved over at least one of theplurality of previously sensed communication conditions, selecting oneof said previous modes as the selected mode to be applied by saidencoder.
 2. The method of claim 1, wherein said values are expressed asCodec Mode Requests (CMRs).
 3. The method of claim 1, wherein saidvalues are expressed as Codec Mode Commands.
 4. The method of claim 1,further comprising the step of dynamically selecting the length M ofsaid memory with respect to the Adaptive Multi-Rate loop delay.
 5. Themethod of claim 4, wherein the length M of said memory is selectedresponsive to said Adaptive Multi-Rate loop delay only in the case wherethe Adaptive Multi-Rate loop delay is higher than a predeterminedthreshold.
 6. The method of claim 1, wherein the length M of said memoryis selected so that, when said first mode is less robust than a modecurrently employed by said encoder, adaptation to said first mode doesnot coincide in time with a degraded communication condition of saidcommunication channel, whereby out of phase conditions in modeadaptation are avoided.
 7. The method of claim 1, wherein the length Mof said memory is selected so that, when said first mode is less robustthan a mode currently applied by said encoder, adaptation to said firstmode is delayed until communication conditions remain acceptable for thefirst mode.
 8. The method of claim 1, further comprising the step ofproviding, by said second node, an indication of the selected mode tothe first node for adaptation of said encoder.
 9. The method of claim 1,the first node being a mobile station and the second node being a basestation.
 10. The method of claim 1, the first node being a base stationand the second node being a mobile station.
 11. A delay unit forcontrolling the operation of a an Adaptive Multi-Rate encoder forencoding speech signals to be sent between a first node and a secondnode in a wireless communications system, the encoder being operable toencode the speech signals in at least two codec modes having differentdegrees of robustness, the delay unit comprising: an input for receivingsignal values representing communication conditions in a communicationchannel between the first node and the second node; a memory for storingthe signal values in memory locations, the memory having a length M ofmemory locations that is selected responsive to an Adaptive Multi-Rateloop delay, M being an integer; a selector coupled to the memorylocations for comparing a last received signal value to a plurality ofpreviously received signal values that are currently stored in thememory locations, said last received signal value corresponding to acurrent communication condition and each said plurality of previouslyreceived signal values corresponding to one of a plurality of previouscommunication conditions wherein, as compared to the plurality ofpreviously received signal values, if the last received signal valueindicates that the current communication condition is degraded relativeto the plurality of previous communication conditions, selecting amemory location containing said last received signal value; or isimproved over at least one of the plurality of previous communicationconditions, selecting a memory location containing one of said pluralityof previously received signal values, the selector making memoryselections according to a predetermined filter algorithm; and a switch,controlled by said selector, for switching the signal value contained inthe selected memory location of the memory to be passed to an output ofthe delay unit.
 12. The delay unit of claim 11 wherein said signalvalues are channel measurement values.
 13. The delay unit of claim 11,wherein said signal values are indicative of codec modes applicable tosaid encoder.
 14. The delay unit of claim 11, wherein the length M ofsaid memory is selected adaptively with respect to the AdaptiveMulti-Rate (AMR) loop delay.
 15. The method of claim 11, wherein thelength M of said memory is selected so that adaptation to a codec modethat is less robust than a codec mode currently applied by said encoderdoes not coincide in time with a degraded communication condition ofsaid communication channel, whereby out of phase conditions in modeadaptation are avoided.
 16. The delay unit of claim 11, wherein thelength M of said memory is selected so that adaptation to a codec modethat is less robust than a codec mode currently applied by said encoderis delayed until communication conditions remain acceptable for the lessrobust codec mode.
 17. The delay unit of claim 11, said memory locationshaving position numbers indicating their sequential order, saidpredetermined filter being operable to sort the signal values stored insaid memory locations according to the quality of the communicationcondition indicated and said selector being operable to select thememory location having position number m smaller than or equal to saidmemory length M containing one of said plurality of previously receivedsignal values.
 18. The delay unit of claim 11, said predetermined filterbeing operable to search among the signal values stored in said memorylocations for a signal value that represents a worst communicationcondition or an n-th worst communication condition, and said selectorbeing operable to select the memory location for the signal value thatrepresents the worst communication condition or the n-th worstcommunication condition as said memory location containing one of saidplurality of previously received signal values.
 19. The delay unit ofclaim 11, said predetermined filter being operable to determine a medianof the signal values stored in said memory locations, and said selectorbeing operable to select the memory location containing the median ofthe signal value as said memory location containing one of saidplurality of previously received signal values.
 20. A method ofcontrolling the operation of an Adaptive Multi-Rate encoder for encodingspeech signals to be sent between a first node and a second node in awireless communication system, the encoder being operable to encode thespeech signals in at least two modes having different degrees ofrobustness, the method comprising the steps of: sensing a firstcommunication condition in a communication channel between the first andthe second node; generating a first channel measurement valuecorresponding to the first communication condition in the communicationchannel; storing the first channel measurement value in a memory, thelength M of which is selected responsive to an Adaptive Multi-Rate loopdelay, where M is an integer; and comparing the first channelmeasurement to a plurality of channel measurement values previouslystored in the memory, each of said plurality of channel measurementvalues corresponding to one of a plurality of previously sensedcommunication conditions wherein, compared to the plurality ofpreviously stored channel measurement values, if the first channelmeasurement value indicates that the first communication condition isdegraded relative to the plurality of previously sensed communicationconditions, selecting a first mode associated with the first channelmeasurement value; is improved over at least one of the plurality ofpreviously sensed communication conditions, selecting a second modeassociated with one of said plurality of channel measurement valuespreviously stored in said memory.
 21. The method of claim 20, furthercomprising the step of dynamically selecting the length M of said memorywith respect to the Adaptive Multi-Rate loop delay.
 22. The method ofclaim 21, wherein the length M of said memory is selected responsive tosaid Adaptive Multi-Rate loop delay only in the case where the AdaptiveMulti-Rate loop delay is higher than a predetermined threshold.
 23. Themethod of claim 20, wherein the length M of said memory is selected sothat, when said first mode is less robust than a mode currently employedby said encoder, adaptation to said first mode does not coincide in timewith a degraded communication condition of said communication channel,whereby out of phase conditions in mode adaptation are avoided.
 24. Themethod of claim 20, wherein the length M of said memory is selected sothat, when said first mode is less robust than a mode currently employedby said encoder, adaptation to said first mode is delayed untilcommunication conditions remain acceptable for the first mode.
 25. Themethod of claim 20, further comprising the step of providing, by saidsecond node, an indication of either the first or the second mode to thefirst node for adaptation of said encoder.
 26. The method of claim 20,the first node being a mobile station and the second node being a basestation.
 27. The method of claim 20, the first node being a base stationand the second node being a mobile station.
 28. A method of controllingthe operation of an Adaptive Multi-Rate encoder for encoding speechsignals to be sent using Tandem Free Operation over a firstcommunication channel and a second communication channel that areconnected in series via a transport network between a first node and asecond node in a wireless communication system, the encoder beingoperable to encode the speech signals in at least two modes havingdifferent degrees of robustness, the method comprising the steps of:sensing a first communication condition in the first communicationchannel between the first node and the transport network; identifying afirst mode applicable to said encoder corresponding to the firstcommunication condition in the first communication channel; storing avalue indicative of the first mode in a memory, the length M of which isselected responsive to an Adaptive Multi-Rate loop delay, where M is aninteger; comparing the value indicative of the first mode to a pluralityof values previously stored in the memory, each of said plurality ofvalues being indicative of one of a set of previous modes correspondingto one of a plurality of previously sensed communication conditionswherein, compared to the plurality of previously stored values, if thevalue indicative of the first mode indicates that the firstcommunication condition is degraded relative to the plurality ofpreviously sensed communication conditions, selecting said first mode asa candidate mode to be applied by said encoder; or is improved over atleast one of the plurality of previously sensed communicationconditions, selecting one of said previous modes as said candidate modeto be applied by said encoder; receiving a value indicative of a secondmode corresponding to a second communication condition in said secondcommunication channel between said transport network and said secondnode; and selecting as a selected mode to be applied by said encoder themost robust one of said candidate mode and said second mode.
 29. Themethod of claim 28, further comprising the step of providing anindication of the selected mode to the first node for adaptation of saidencoder.
 30. An arrangement operable to control the operation of anAdaptive Multi-Rate encoder operable to encode speech signals to be sentusing Tandem Free Operation over a first communication channel and asecond communication channel that can be connected in series via atransport network between a first node and a second node in a wirelesscommunication system, the encoder being operable to encode the speechsignals in at least two modes having different degrees of robustness,the arrangement comprising: an analyzer operable to sense communicationconditions in the first communication channel between the first node andthe transport network; a selector operable to identify modes applicableto said encoder corresponding to the communication conditions in thefirst communication channel; a memory for storing values indicative ofmodes applicable to said encoder, the length M of the memory beingselectable responsive to an Adaptive Multi-Rate loop delay, where M isan integer; a filter unit operable to compare a first value indicativeof a first mode corresponding to a first communication condition in thefirst communication channel to a plurality of values previously storedin the memory, each of said plurality of values being indicative of oneof a set of previous modes corresponding to one of a plurality ofpreviously sensed communication conditions in the first communicationchannel wherein, compared to the plurality of previously stored values,if the first value indicative of the first mode indicates that the firstcommunication condition is degraded relative to the plurality ofpreviously sensed communication conditions, the filter unit is operableto select said first mode as a candidate mode to be applied by saidencoder; or is improved over at least one of the plurality of previouslysensed communication conditions, the filter unit is operable to selectone of said previous modes as said candidate mode to be applied by saidencoder; input means operable to receive a value indicative of a secondmode corresponding to a second communication condition in said secondcommunication channel; a combiner operable to select as a selected modeto be applied by said encoder the most robust one of said candidate modeand said second mode; and output means operable to provide an indicationof the selected mode to the first node for adaptation of said encoder.31. An arrangement operable to control the operation of an AdaptiveMulti-Rate encoder operable to encode speech signals to be sent usingTandem Free Operation over a first communication channel and a secondcommunication channel that can be connected in series via a transportnetwork between a first node and a second node in a wirelesscommunication system, the encoder being operable to encode the speechsignals in at least two modes having different degrees of robustness,the arrangement comprising: a first input for receiving signal valuesrepresenting communication conditions in the first communicationschannel between the first node and the transport network; a memory forstoring the signal values in memory locations, the memory having alength M of memory locations that is selected responsive to an AdaptiveMulti-Rate loop delay, M being an integer; a selector coupled to thememory locations for comparing a last received signal value to aplurality of previously received signal values that are currently storedin the memory locations, said last received signal value correspondingto a current communication condition in the first communication channeland each said plurality of previously received signal valuescorresponding to one of a plurality of previous communication conditionsin the first communication channel wherein, as compared to the pluralityof previously received signal values, if the last received signal valueindicates that the current communication condition is degraded relativeto the plurality of previous communication conditions, selecting amemory location containing said last received signal value; or isimproved over at least one of the plurality of previous communicationconditions, selecting a memory location containing one of said pluralityof previously received signal values, the selector making memoryselections according to a predetermined filter algorithm; and a switch,controlled by said selector, for switching the signal value contained inthe selected memory location of the memory to be passed to a firstoutput, said signal value contained in the selected memory locationbeing indicative of a candidate mode to be applied by said encoder; asecond input for receiving a value indicative of a second modecorresponding to a second communication condition in said secondcommunication channel; a combiner operable to select as a selected modeto be applied by said encoder the most robust one of said candidate modeand said second mode; and a second output for providing an indication ofthe selected mode or a value indicative of the selected mode to thefirst node for adaptation of said encoder.